Recently, developments concerning communication networks focus also on a communication network enabling transmission of (speech) data on the basis of the Internet Protocol (IP), known as Voice over IP (VoIP) and/or IP Telephony.
Such communication networks comprise an access network via which a subscriber's terminal gains access and/or connects to a core network. The access network is specific to a connection technology such as wireless/radio access networks or non-wireless access networks, while the core network is independent of the connection technology adopted for connection of the terminal to the network.
FIG. 1 of the drawings shows in rough outline the basic constituents of such a communication network. It is to be noted that the drawing illustration has been significantly simplified for explanatory purposes such that FIG. 1 shows only those components considered to be relevant for the understanding of the present invention.
A subscriber, i.e. a subscriber's terminal A establishes a communication path via the network NW to a communication destination B. The subscriber's terminal may be any wireless or non-wireless terminal, as long as it is adapted to co-operate with the communication network NW. Likewise, the communication destination may be a terminal of a subscriber to said network. Nevertheless, it may be any terminal and also a terminal having subscribed to another network (not shown). A terminal may be identified by its address or an address associated thereto.
The communication network NW consists of an access network AN and a core network CN. Since the access network technology is of minor interest for the present invention, a detailed description thereof is omitted. However, the access network may for example be a UMTS (Universal Mobile Telecommunications System) access network, a GPRS (General Packet Radio Service) access network, or the like.
The core network CN in turn comprises at least one call control entity also referred to as call state control functional entity CSCF (or referred to as service switching point SSP). The call state control function CSCF comprises a serving CSCF and an interrogating CSCF. The serving CSCF is used for mobile originated communications and also to support mobile terminated communications. The serving CSCF provides a serving profile database SPD (not shown) and an address handling AH (also not shown) functionality. The serving CSCF supports the signaling interactions with a terminal. The interrogating CSCF is used for mobile terminated communications and is also used to determine how to route mobile terminated calls. The interrogating CSCF interrogates a home subscriber server HSS (not shown) for information to enable the call to be directed to the serving CSCF. The interrogating CSCF provides an incoming call gateway functionality ICGW and the address handling AH functionality. Both, the serving and interrogating CSCF components can be provided in a single CSCF if required.
Particularly, the call state control function CSCF provides a call control functionality CCF. In connection therewith, it provides among others for                call set-up/termination and state/event management        call event reporting for billing, auditing, intercept or other purposes,        receiving and processing application level registrations.        
Furthermore, the call state control functional entity CSCF has an interface to a service control entity also referred to as service control point SCP. This network entity provides for the ability of the network to determine what a particular service does, for a specific invocation of that service, within the limitations of that service.
Thus, in brief, a subscriber's terminal A, upon establishing a call, communicates via the access network, the CSCF of the core network, and again via the access network with the communication destination B (terminal of subscriber B, for example), while the services provided for such a call are under control of the SCP.
In currently developed communication networks, signaling within a network as described above is based on the Session Initiation Protocol SIP. According to the Session Initiation Protocol, SIP message headers are in plain text and look similar to e-mails. Also, SIP uses a client server model similar to the Hypertext transfer protocol HTTP and many others, and is used in conjunction with other protocols such as Session Description Protocol SDP etc.
Signaling is effected by exchanging SIP messages. SIP messages can be divided into SIP request messages and SIP response messages.
SIP request messages are SIP INVITE (invites client or server to establish a session), SIP ACK (confirmation reception of a final response to an INVITE message), SIP BYE (sender wishes to close the session), SIP CANCEL (cancels pending requests), SIP OPTIONS (asks for information about capabilities before establishing a session), SIP REGISTER (informs a location server of the client's IP address).
SIP response messages are 1xx Informational, 2xx Success, 3xx REDIRECTION, 4xx CLIENT ERROR, 5XX SERVER ERROR, 6XX GLOBAL FAILURE. Such SIP response messages use a 3 digit number, e.g. 1xx. The first digit defines the category, while the next two digits allow up to 100 variations, e.g. 200 OK (successful invitation).
The call state control functional entity CSCF in connection with the SIP serves as a SIP server with which session establishment is effected according to known SIP session establishment procedures which are not described here in detail. A SIP server similar to a CSCF is a so-called call processing server CPS (not shown in FIG. 1). The difference being that a CPS is compliant to SIP (as defined in RFC 2543), while the CSCF is compliant to 3GPP specifications. It can be said that a CSCF is a subclass of call processing servers designed for mobile networks.
The above briefly described SIP standards enable the sending of different types of content within the SIP signaling, for example with the SIP INVITE message. The content is defined with content type and content length field within the SIP message, whereas the actual content is inserted into the message body. Such content to be inserted can be any content which has a MIME type definition. MIME (=Multipurpose Internet Mail Extensions) provide a common structure for encoding a range of electronic documents or other multimedia data. For example, such MIME type content could be text, image, audio, video, compressed file, application data, HyperText Markup Language (HTML) data, e-mail data.
Using MIME enables signaling the type of “object” being carried, associating a file name with an object, associating several independent objects for forming a multi-part object, handling data encoded in text or binary and—if necessary— re-encoding the binary as text.
However, there may arise situations in which there are conditions prevailing in the network which impede the establishment of a call connection between a caller and a called destination (callee) and which will lead to missed calls.
In order to provide data representing information concerning missed calls to the callee, call answering services as a basic value added service in today's Public Land Mobile Networks (PLMNs) have been proposed. Such services are normally implemented using a voice mail system in connection with a call transfer function of the PLMN. In practice, a call answering service is often used for a mere recording of call/callback requests (by leaving a message such as “could you please call me back under telephone number 12345”).
A call answering service is not an optimal service for call request logging although it is used for that purpose. The user (callee) must complete several phases before being able to call back the caller waiting for being called back. Namely, the callee will have to    a) receive a notification (e.g. by SMS, Short Message Service) that a voice mail has arrived for his (mobile) terminal,    b) call his voice mail box,    c) listen to the voice mail and note down the telephone number he is asked to call back,    d) terminate the call to the voice mail box    e) make a new call to the phone number obtained under step c).
Missed calls are logged in today's mobile phones (terminals) but such a phone service cannot be used extensively because there exist limitations such as    1. mobile phones can not be always switched on (because of being present in or near airplanes/hospitals etc. where there might be a danger of adverse effects due to the radio waves emitted by the mobile terminal,    2. a caller is not informed that his failed call attempt has been logged in the called phone, which often means that the caller will additionally leave a voice message using the call answering service just requesting to be called back, and    3. it is difficult for the called person to derive a knowledge of who was the caller of the logged phone number (CLI, Calling Line Identity) if the logged CLI is not registered to the phonebook of his terminal (i.e. mobile phone).
Still further, with the use of SIP messages (as described above) such as SIP INVITE for call establishment, even in case of leaving a voice message at an answering service, content already carried in the SIP message is not known to the callee and would thus have to be retransmitted later on, leading to a wasteful use of network transmission capacity.